1. Field of the Invention
The invention relates to electronic communication and, more particularly, to wired and cordless telephone systems.
2. Description of the Related Art
Telephone systems have evolved with the central goal of transmitting human-voice signals. Other goals have been added with the increasing capacities of telephone systems and the increasing demands of telephone users. A major change has been the rapid growth of digital telephone links for long-distance and local connections. With this growth has come the expectation of increased bandwidth on a telephone connection.
To transmit a voice signal or other analog signal through a digital link, the analog signal is digitized so that it can be represented by a stream of information symbols. Digitizing the audio signal involves sampling it so that values are recorded only at discrete points in time. Each sampled signal is subsequently quantized so that its amplitude is recorded as one of a discrete set of possible values. For human voice, the spectral power distribution has a bandwidth of approximately 3 kHz, so a sampling rate of 8000 samples/second (8 kS/s) records sufficiently many samples to reproduce the signal. With this sampling rate, 256 appropriately-chosen quantization levels (such as the logarithmically spaced A-law or xcexc-law levels) are sufficient for producing a xe2x80x9ctoll-qualityxe2x80x9d digital audio signal.
Within the nominal 3 kHz bandwidth, the typical spectral power distribution of human voice is highly peaked in the 200-800 Hz frequency range, with diminishing amplitudes at higher frequencies. Since the power spectral density of a voice signal is not flat over the nominal 3 kHz frequency range, a digitized voice signal can be compressed and decompressed without much loss of fidelity. This compression may be performed in conjunction with the digitization, by using different forms of differential pulse code modulation (DPCM) such as delta modulation, xe2x80x9clinear delta modxe2x80x9d (LDM), continuously variable slope delta modulation (CVSD), or various forms of adaptive differential pulse code modulation (ADPCM). Because of their reduced data rates, these coding schemes are commonly used in many communications systems for voice signals. These coding schemes and others are used in many implementations of telephone equipment. While these schemes work well for voice signals, they are not adequate for high quality audio signals. Indeed, many aspects of telephone equipment make standard telephones unsuitable for high quality audio. Even without the voice-oriented compression features (ADPCM, CVSD, etc.), regular telephones are not desirable for communicating clear audio over the audible frequency spectrum. The primary limitation comes from a systemic design consideration: each call is generally limited to a roughly 3 kHz bandwidth.
Telephone subscribers communicate via a vast telephone network, referred to as the Public Switched Telephone Network (PSTN). In the present disclosure, the terms xe2x80x9cPSTNxe2x80x9d and xe2x80x9ctelephone linexe2x80x9d are intended to include the analog or POTS (Plain Old Telephone Service), PBX (Private Branch Exchange), ISDN (Integrated Services Digital Network), DSL (Digital Subscriber Line), and Wireless Local Loop (WLL), among others. In describing the methods and systems that apply to a xe2x80x9ctelephone,xe2x80x9d it is noted that these methods and systems can be used in a standard telephone, or in other telephone devices configured or adapted to communicate on these networks, such as personal computers and household appliances with remote-access functionality. A telephone subscriber""s communications devices, e.g., telephones, are typically connected in parallel to a telephone line that links a subscriber""s premises to a telephone service provider""s central office (CO). The communications link is generally carried from the caller to the receiving party through one or more CO""s. The links from caller to CO and from CO to receiving party are generally made over twisted-pair copper lines, which have a wide bandwidth (10 MHz-10 GHz). In a few situations therefore, a caller and a call recipient may communicate over a high-bandwidth channel. In general, however, the transmitted signal is filtered to have roughly a 3 kHz bandwidth adequate for most human-voice communication. This filtering, performed either at the CO or at a switching unit located near the caller, makes efficient use the PSTN""s switching equipment and its multiplexed communications lines. The filtering is especially important in the case of telephone calls involving more than one CO, such as xe2x80x9clong-distancexe2x80x9d calls. In this case, the communications link between CO""s is typically carried over digital communications channels, such as fiber-optic networks and satellite relays. The digital links are generally designed to accommodate only a 64 kbps data rate (=8 bits/Samplexc3x978000 Samples/second) for each telephone call, which limits the bandwidth of the telephone signal to roughly 3 kHz. Since essentially all long distance and local voice-traffic calls are limited by the design of the PSTN to the 3 kHz bandwidth, telephone equipment is generally made with this design limitation as well. Thus, speakers, microphones, and circuitry in most telephones are generally optimized for the main frequency range of the human voice, nominally 30 Hz-3 kHz. This makes most telephone equipment unsuitable for transmission of high quality audio signals. CD-quality music, for example, with stereophonic channels and an approximate frequency range of 20 kHz, is transmitted poorly through telephone equipment.
Like regular wired telephones, cordless telephones generally limit the bandwidth of the audio signals they carry. Cordless telephones use either analog or digital links between a handset and a base unit. These units generally communicate through a wireless link such as a radio (although the signal may alternatively be transmitted through a free-space optical signal, a waveguide, or an optical fiber). In general, the voice data on a digital wireless link are compressed through techniques such as ADPCM or CVSD coding. The wireless link thus only needs to carry payload data at a reduced rate, typically 30%-60% less than the nominal 64 kbps rate.
Described herein is a telephone that communicates high-quality audio signals, such as xe2x80x9cCD-qualityxe2x80x9d sound (sampled, for example, at 44.1 kHz with 16-bit resolution). In one embodiment, the telephone includes a sampler, a compression block, a telephone port, a decompression block, and a digital-to-analog (D/A) converter.
The sampler receives an analog audio signal (generated, for example, by a microphone or supplied through an audio port in the telephone) and digitizes the signal into a digital audio signal with a high bit rate (such as 44.1 kSamples/secxc3x9716 bits/Sample=705.6 kbps) that permits the communication of high-quality audio. The compression block then encodes the digital audio signal into a compressed digital signal at a lower bit rate (such as 150 kbps or 56 kbps). The compressed digital signal is communicated through the telephone port to a telephone network.
An incoming compressed digital signal is received from the telephone network by the telephone port and provided to the decompression block. From the received compressed digital signal, the decompression block reconstructs a digital audio signal at the high data rate. The D/A converter generates a reconstructed analog audio signal in response to the reconstructed digital audio signal.
Also described herein is a method for communicating an audio signal with an extended frequency range over a telephone network. The method preferably includes steps of sampling the audio signal to generate a digitized signal, compressing the digitized signal with a compression algorithm, communicating the compressed signal over the telephone network, decompressing the compressed signal to generate a recovered digitized signal, and converting the recovered digitized signal to a recovered analog audio signal. The compression algorithm preferably includes psycho-acoustic perceptive coding and preferably generates an MPEG-audio data stream.